My ffmpeg code. The output produced, with a WAV header for the sample rate of the original file, sounds like a sped up and distorted version of the original.

ffmpeg-test.c

#include <libavformat/avformat.h>
#include <libavutil/samplefmt.h>
#include <string.h>
#include <stdio.h>

const char* get_str_err(const int err) {
	static char buf[1024];
	if (av_strerror(err, buf, sizeof(buf))) {
		strcpy(buf, "Couldn't get a proper error!");
	}
	return buf;
}

int main(int argc, char **argv) {
	int err;

	av_register_all();

	// setup format context
	AVFormatContext *fctx = NULL;
	if ((err = avformat_open_input(&fctx, argv[1], NULL, NULL)) < 0) {
		fprintf(stderr, "Couldn't open input: %s\n", get_str_err(err));
		return err;
	}

	if ((err = avformat_find_stream_info(fctx, NULL)) < 0) {
		fprintf(stderr, "Couldn't find stream info: %s\n", get_str_err(err));
		return err;
	}

	// find codec and stream ID
	AVCodec *codec = NULL;

	int stream_id = av_find_best_stream(fctx, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
	fprintf(stderr, "Stream ID: %i\n", stream_id);

	AVStream *stream = NULL;
	stream = fctx->streams[stream_id];

	AVCodecContext *codec_ctx = NULL;
	codec_ctx = stream->codec;

	// setup codec context
	if ((err = avcodec_open2(codec_ctx, codec, NULL)) < 0) {
		fprintf(stderr, "Couldn't open codec: %s\n", get_str_err(err));
		return err;
	}

	AVFrame *frame = av_frame_alloc();
	AVPacket packet;
	int got_frame = 0;

	// decode frames from file and output to stdout
	for (;;) {
		err = av_read_frame(fctx, &packet);
		if (err == 0xDFB9B0BB) {break;} // EOF
		if (packet.stream_index != stream_id) {continue;}
		int total_length = 0;

		for (;;) {
			int len = avcodec_decode_audio4(codec_ctx, frame, &got_frame, &packet);
			if (len == 0) {continue;} //skip if nothing was decoded

			if (got_frame) {
				int data_size = av_samples_get_buffer_size(NULL, codec_ctx->channels, frame->nb_samples, codec_ctx->sample_fmt, 1);
				fwrite(frame->data[0], 1, data_size, stdout);
			}
			total_length += len;
			if (total_length >= packet.size) {
                break;
            }
		}
	}

	//flush decoder
	for (;;) {
		packet.data = NULL;
		packet.size = 0;
		int len = avcodec_decode_audio4(codec_ctx, frame, &got_frame, &packet);
		if (len == 0) {break;} //nothing left

		if (got_frame) {
			int data_size = av_samples_get_buffer_size(NULL, codec_ctx->channels, frame->nb_samples, codec_ctx->sample_fmt, 1);
			fwrite(frame->data[0], 1, data_size, stdout);
		}
	}

	return 0;
}